1. Field of the Invention
The present invention relates generally to improvements in voice communication systems and more particularly pertains a new and improved method and apparatus for interfacing a public switched telephone network (PSTN) with an Internet Protocol (IP) network to provide real-time voice communication and messaging services over the two networks.
2. Description of the Related Art
Voice communication over Internet Protocol networks has been accomplished by using computers with sound cards to communicate with other computers with sound cards connected to the IP network through a service provider. Such devices have been unsatisfactory in that the PSTN is excluded from the communication loop.
Attempts have been made to integrate the public switched telephone network into the Internet Protocol network. Such attempts have been limited to messaging systems such as described in U.S. Pat. No. 5,608,786, granted Mar. 4, 1997, to Allistair T. Gordon for A UNIFIED MESSAGING SYSTEM AND METHOD, the disclosure of which is incorporated herein by reference. Although the system and method described in this '786 patent utilize both the Internet Protocol network and public switch telephone network, it does not provide for real-time voice communication between units connected to the Internet Protocol network and the public switch telephone network.
Voice technology, which has been around for over one hundred years, has been evolving since the first phone call was made. The standard public switched telephone network (PSTN) which is basically a large circuit-switched network, is truly ubiquitous, simple to use, dependable and pervasive.
Voice technology today involves both analog and digital transmission and signaling. Human speech and everything we hear is in analog form. The telephone network was based upon an analog infrastructure. Thus, early analog phones utilized a carbon microphone, a battery, an electromagnet and an iron diaphragm. Connecting these components together produced a method of transporting voice. Although analog communication is ideal for human communication, it is neither robust nor an efficient method of transmitting information.
Digital transmission of information is much more desirable. Digital samples comprise one and zero bits. It is much easier to separate digital samples from line noise. Thus, when digital signals are regenerated, a clean sound can be maintained. As a result of the benefits of digital representation of the analog voice signals, pulse code modulation techniques were integrated into the telephone network. Pulse code modulation (PCM) converts analog sound into digital form by sampling the analog sound so many times per second and converting the sound into a numeric code. After the analog wave form is sampled, it is converted into a discrete digital form, as samples represented by code that indicates the amplitude of the wave form at the instant the sample was taken. A standard telephone form of PCM uses 8 bits for the code and a logarithm compression method that assigns more bits to lower amplitude signals. A standard transmission rate of 64K bits per second is used for one channel of telephone digital communication. The two basic variations of 64K bps PCM are μ-law and A-law. Both methods are similar in that they both use logarithmic compression to achieve 12-13 bits of linear PCM quality with 8 bits. They differ in relatively minor compression details. North America uses μ-law modulation. Europe uses A-law modulation. Another compression method that is often used today is an adaptive differential pulse-code modulation (ADPCM). A commonly used form of ADPCM is ITU-T G.726, which encodes by using 4 bit samples giving a transmission rate of 32K bps Unlike PCM, the 4 bits do not directly encode the amplitude of speech, but rather the differences in amplitude as well as the rate of change of that amplitude employing rudimentary linear prediction.
Both PCM and ADPCM are examples of wave form coder-decoders (CODECs), compression techniques that exploit redundant characteristics of the wave form itself. Many variations of CODEC compression techniques have been suggested, some of which have been written into standards promulgated by the ITU-T in its G-series recommendations, for example, such as G.711, G726, G728, G729, and G723.1.
Although these compression techniques seem to have successfully addressed the problem of noise on the propagation medium, delay is still a major consideration in today's telephony networks. There are basically two types of delay, propagation delay and handling delay. Propagation delay is caused by the speed of light in a fiber or copper based network. Handling delay is caused by devices that handle the voice information along the voice path. The speed of light in a vacuum is 186,000 miles per second. Electrons travel 100,000 miles per second in copper. A fiber network half way around the world (13,000 miles) only induces a one way delay of about 70 milliseconds. Such a delay is almost imperceptible to the human ear. But these propagation delays in combination with handling delays can cause noticeable speech degradation. Handling delays become a large issue in packetized environments, which are utilized in Internet Protocol networks. A typical packetizer such as made by Cisco Systems, for example, generates a speech frame every 10 milliseconds. Two of these speech frames are then placed into one packet and a real-time transport protocol header is then attached to the packet.
Another problem experienced in traditional toll networks is echo. Echo is normally caused by mismatch in impedance between the 4-wire network switch conversion to a 2-wire local loop. Although hearing your own voice in the receiver is common and reassuring to a speaker, hearing your own voice in a receiver longer then 2.5 milliseconds will cause interruptions and breaks in the conversation. As a result, echo in the standard PSTN is controlled with echo cancelers and a tight control on impedance mismatches at the common reflection points. In packet based networks, echo cancelers are built into the low bit rate CODECS.
Various types of in-band and out of band signaling methods are used in today's telecommunication networks. A common method of in-band signaling is the use of single or multi-frequency tones. A common method of out of band signaling is integrated services digital network (ISDN) which used the D channel for call set up. Out of band signaling is what it says. It uses a separate channel for signaling outside the voice band.
Facsimile machines that are commonly used today implement ITU recommended protocols T.30 and T.4. The T.30 protocol describes the formatting of non-page data such as messages that are used for capabilities and negotiation. The T.4 protocol describes formatting of page image data.
In a PSTN, the fax machines synchronize their transmissions end to end and negotiate page by page. In a packet-based network like in an IP network, the T.30 protocol engines are de-coupled and demodulated, allowing for delays inherent in the network.
Another ITU-T specification of considerable importance is H.323 which is utilized for transmitting multimedia (voice, video, and data) across a local area network which can be an IP network or a network of any other protocol. H.323 describes H.323 terminals, 11.323 MCUs, 11.323 gateways, and H.323 gatekeepers. An H.323 gatekeeper for example, performs address translation, admission control, bandwidth management and zone management. An H.323 gateway provides a gate between an IP protocol network and the PSTN as well as any other H.320 terminals, V.70 terminal, H.324 terminal, and any other speech terminals. The H.323 protocol is used for audio, video and data applications and system control.
Packet voice applications readily lend themselves to transmitting voice over IP networks, thus presenting a fundamental change in the PSTN approach of offering telephony services. One of the main reasons packet telephony has been gaining interest is the cost saving available. By integrating the voice and the data networks into one network considerable cost savings can be achieved. A voice over IP network permits toll bypass which allows the customers to replace their tie lines that currently hook up their PBX to PBX networks, and route their voice calls across their existing data structure utilizing the IP network.
Turning now to FIG. 1, use of the present invention in seamlessly merging a PSTN and IP network for voice communication is illustrated. The global Internet system 13 is an Internet Protocol (IP) network. To use this IP network a subscriber typically contracts with a commercial access or service provider, obtains an Internet address and the capability to thereby send and receive e-mail by way of the IP network, and perform other functions supported by the IP network. The subscriber typically uses a personal computer and modem to contact the service provider over a public switched telephone network or any other convenient communication link such as cable or DSP line. Once connected to the IP network 13 the subscriber may communicate with any other subscribers connected to the network, which subscribers may be located in a host of different countries.
Local PSTN networks 15 and 17 exist throughout the United States and throughout the world. These networks are administered by local and long distance telephone companies. Access to the local PSTN networks 15 and 17 is also by contract between a subscriber and the PSTN and the local telephone company operating the PSTN. Typically the local PSTNs are connected over long distance trunks 18, which may consist of anything from wire lines and optical fiber to wireless satellite links.
A typical PSTN 15 would interconnect a plurality of phones 39, 27 by wireline connections 47 and 31 respectively, a plurality of faxes 41 by wireline connections 49, and perhaps a wireless communication network 29 by way of trunk lines 33. The wireless communication network 29 would communicate with a plurality of cell phones 35 and pagers 37 over wireless links 43 and 45.
In addition to these units, a local PSTN 17 for example, besides connecting standard telephone sets 93 over wire lines 97 and faxes 91 over wire lines 95 may connect to private branch exchange (PBX) units 71 over trunk lines 73. The private branch exchange 71 is typically located at a business sight. It would connect a plurality of telephone units 75, 77 and 79 over wire lines 81, 83 and 89 to the local PSTN 17.
All the units connected to the local PSTNs 17 and 15 are capable of communicating with any other units connected to these PSTNs because the local PSTNs are in turn connected together by trunk lines 18. This is a traditional telephone network.
The IP network 13 is designed for interconnecting computers for communication purposes. Access to the IP network 13 is through a service provider. A typical subscriber like computer 59 for example, would connect to the IP network 13 over a connecting link 61 which may consist of a modem and local telephone line, a digital cable or other means commonly available. The computer subscriber 59 typically pays a monthly access fee to the service provider. Communication between the subscribers to the IP network is usually by e-mail.
Computers with multimedia capability and a voice packetizing program such as Netmeeting for example which is a software program available on the Internet at no charge can communicate with other computers having multi-media and net media capability by voice signaling. Thus computer 63 which has multimedia capability and Netmeeting software could take the voice signals from a phone 65 which is connected by a wireline 67 to computer 63, packetize it into a digital format and transmit it over the modem or cable or DSP line 69 to the IP network 13 where it would be distributed to computer 59, for example, and broken down in computer 59 to a voice message. This system only provides voice communication over the IP network between devices connected to the IP network 13.